[asterisk-commits] dvossel: branch 1.6.2 r196452 - in /branches/1.6.2: ./ channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri May 22 16:59:38 CDT 2009
Author: dvossel
Date: Fri May 22 16:59:31 2009
New Revision: 196452
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=196452
Log:
Merged revisions 196416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
SIP set outbound transport type from Registration
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
........
Modified:
branches/1.6.2/ (props changed)
branches/1.6.2/channels/chan_sip.c
branches/1.6.2/configs/sip.conf.sample
Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=196452&r1=196451&r2=196452
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Fri May 22 16:59:31 2009
@@ -1845,6 +1845,8 @@
AST_STRING_FIELD(useragent); /*!< User agent in SIP request (saved from registration) */
);
struct sip_socket socket; /*!< Socket used for this peer */
+ enum sip_transport default_outbound_transport; /*!< Peer Registration may change the default outbound transport.
+ If register expires, default should be reset. to this value */
unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
struct sip_auth *auth; /*!< Realm authentication list */
int amaflags; /*!< AMA Flags (for billing) */
@@ -1859,7 +1861,7 @@
int lastmsgssent;
unsigned int sipoptions; /*!< Supported SIP options */
struct ast_flags flags[2]; /*!< SIP_ flags */
-
+
/*! Mailboxes that this peer cares about */
AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;
@@ -1881,7 +1883,7 @@
struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
struct sockaddr_in addr; /*!< IP address of peer */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
-
+
/* Qualification */
struct sip_pvt *call; /*!< Call pointer */
int pokeexpire; /*!< When to expire poke (qualify= checking) */
@@ -3198,6 +3200,28 @@
return sip_debug_test_addr(sip_real_dst(p));
}
+/*! \brief Return int representing a bit field of transport types found in const char *transport */
+static int get_transport_str2enum(const char *transport)
+{
+ int res = 0;
+
+ if (ast_strlen_zero(transport)) {
+ return res;
+ }
+
+ if (!strcasecmp(transport, "udp")) {
+ res |= SIP_TRANSPORT_UDP;
+ }
+ if (!strcasecmp(transport, "tcp")) {
+ res |= SIP_TRANSPORT_TCP;
+ }
+ if (!strcasecmp(transport, "tls")) {
+ res |= SIP_TRANSPORT_TLS;
+ }
+
+ return res;
+}
+
/*! \brief Return configuration of transports for a device */
static inline const char *get_transport_list(unsigned int transports) {
switch (transports) {
@@ -4014,10 +4038,9 @@
* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
* \endverbatim
*
- * \todo This function needs to look for ;transport= too
*/
static int parse_uri(char *uri, char *scheme,
- char **ret_name, char **pass, char **domain, char **port, char **options)
+ char **ret_name, char **pass, char **domain, char **port, char **options, char **transport)
{
char *name = NULL;
int error = 0;
@@ -4036,6 +4059,17 @@
error = -1;
}
}
+ if (transport) {
+ char *t, *type = "";
+ *transport = "";
+ if ((t = strstr(uri, "transport="))) {
+ strsep(&t, "=");
+ if ((type = strsep(&t, ";"))) {
+ *transport = type;
+ }
+ }
+ }
+
if (!domain) {
/* if we don't want to split around domain, keep everything as a name,
* so we need to do nothing here, except remember why.
@@ -11137,11 +11171,24 @@
}
}
+static void set_peer_transport(struct sip_peer *peer, int transport)
+{
+ /* if the transport type changes, clear all socket data */
+ if (peer->socket.type != transport) {
+ peer->socket.type = transport;
+ peer->socket.fd = -1;
+ if (peer->socket.tcptls_session) {
+ ao2_ref(peer->socket.tcptls_session, -1);
+ peer->socket.tcptls_session = NULL;
+ }
+ }
+}
+
/*! \brief Expire registration of SIP peer */
static int expire_register(const void *data)
{
struct sip_peer *peer = (struct sip_peer *)data;
-
+
if (!peer) /* Hmmm. We have no peer. Weird. */
return 0;
@@ -11149,7 +11196,8 @@
memset(&peer->addr, 0, sizeof(peer->addr));
destroy_association(peer); /* remove registration data from storage */
-
+ set_peer_transport(peer, peer->default_outbound_transport);
+
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
register_peer_exten(peer, FALSE); /* Remove regexten */
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
@@ -11298,16 +11346,16 @@
We still need to be able to send to the remote agent through the proxy.
*/
if (tcp) {
- if (!parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
+ if (!parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL, NULL)) {
use_tls = TRUE;
} else {
- if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL))
+ if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
/*! \todo XXX why are we setting TLS port if there's no port given? parse_uri needs to return the transport. */
} else {
- if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
+ if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
}
@@ -11344,20 +11392,21 @@
return __set_address_from_contact(pvt->fullcontact, &pvt->sa, pvt->socket.type == SIP_TRANSPORT_TLS ? 1 : 0);
}
-
/*! \brief Parse contact header and save registration (peer registration) */
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
{
- char contact[SIPBUFSIZE];
+ char contact[SIPBUFSIZE];
char data[SIPBUFSIZE];
const char *expires = get_header(req, "Expires");
int expire = atoi(expires);
- char *curi, *host, *pt, *curi2;
+ char *curi, *host, *pt, *curi2, *transport;
int port;
+ int transport_type;
const char *useragent;
struct hostent *hp;
struct ast_hostent ahp;
struct sockaddr_in oldsin, testsin;
+
ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
@@ -11373,8 +11422,6 @@
}
}
- if (peer->socket.type == req->socket.type)
- copy_socket_data(&peer->socket, &req->socket);
copy_socket_data(&pvt->socket, &req->socket);
/* Look for brackets */
@@ -11396,12 +11443,13 @@
} else if (!strcasecmp(curi, "*") || !expire) { /* Unregister this peer */
/* This means remove all registrations and return OK */
memset(&peer->addr, 0, sizeof(peer->addr));
+ set_peer_transport(peer, peer->default_outbound_transport);
AST_SCHED_DEL_UNREF(sched, peer->expire,
unref_peer(peer, "remove register expire ref"));
destroy_association(peer);
-
+
register_peer_exten(peer, FALSE); /* Remove extension from regexten= setting in sip.conf */
ast_string_field_set(peer, fullcontact, "");
ast_string_field_set(peer, useragent, "");
@@ -11422,22 +11470,35 @@
ast_string_field_build(pvt, our_contact, "<%s>", curi);
/* Make sure it's a SIP URL */
- /*! \todo This code assumes that the Contact is using the same transport as the
- REGISTER request. That might not be true at all. You can receive
- sips: requests over any transport. Needs to be fixed.
- Does not parse the ;transport uri parameter at this point, which might be handy
- in some situations.
- */
if (pvt->socket.type == SIP_TRANSPORT_TLS) {
- if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL)) {
- if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL))
+ if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL, &transport)) {
+ if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL, &transport))
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
} else {
- if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL))
+ if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL, &transport))
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
+ }
+
+ /* handle the transport type specified in Contact header. */
+ if ((transport_type = get_transport_str2enum(transport))) {
+ /* if the port is not specified but the transport is, make sure to set the
+ * default port to match the specified transport. This may or may not be the
+ * same transport used by the pvt struct for the Register dialog. */
+ if (ast_strlen_zero(pt)) {
+ port = (transport_type == SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
+ }
+ } else {
+ transport_type = pvt->socket.type;
+ }
+
+ /* if the peer's socket type is different than the Registration
+ * transport type, change it. If it got this far, it is a
+ * supported type, but check just in case */
+ if ((peer->socket.type != transport_type) && (peer->transports & transport_type)) {
+ set_peer_transport(peer, transport_type);
}
oldsin = peer->addr;
@@ -11478,6 +11539,16 @@
peer->addr = pvt->recv;
}
+ /* if the Contact header information copied into peer->addr matches the
+ * received address, and the transport types are the same, then copy socket
+ * data into the peer struct */
+ if ((peer->socket.type == pvt->socket.type) &&
+ (peer->addr.sin_addr.s_addr == pvt->recv.sin_addr.s_addr) &&
+ (peer->addr.sin_port == pvt->recv.sin_port)){
+
+ copy_socket_data(&peer->socket, &pvt->socket);
+ }
+
/* Now that our address has been updated put ourselves back into the container for lookups */
ao2_t_link(peers_by_ip, peer, "ao2_link into peers_by_ip table");
@@ -11497,7 +11568,7 @@
if (peer->is_realtime && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
peer->expire = -1;
} else {
- peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register,
+ peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register,
ref_peer(peer, "add registration ref"));
if (peer->expire == -1) {
unref_peer(peer, "remote registration ref");
@@ -11509,7 +11580,7 @@
XXX WHY???? XXX
\todo Fix this immediately.
*/
- if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
+ if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
ast_db_put("SIP/Registry", peer->name, data);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPort: %d\r\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));
@@ -13215,12 +13286,12 @@
/*! \todo Samme logical error as in many places above. Need a generic function for this.
*/
if (p->socket.type == SIP_TRANSPORT_TLS) {
- if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy)) {
- if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy))
+ if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy, NULL)) {
+ if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy, NULL))
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
} else {
- if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy))
+ if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy, NULL))
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
@@ -22535,9 +22606,9 @@
{
struct sip_peer *peer = NULL;
struct ast_ha *oldha = NULL;
- int found=0;
- int firstpass=1;
- int format=0; /* Ama flags */
+ int found = 0;
+ int firstpass = 1;
+ int format = 0; /* Ama flags */
time_t regseconds = 0;
struct ast_flags peerflags[2] = {{(0)}};
struct ast_flags mask[2] = {{(0)}};
@@ -22600,8 +22671,8 @@
/* If we have realm authentication information, remove them (reload) */
clear_realm_authentication(peer->auth);
peer->auth = NULL;
+ peer->default_outbound_transport = 0;
peer->transports = 0;
- peer->socket.type = 0;
for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
if (handle_common_options(&peerflags[0], &mask[0], v))
@@ -22613,7 +22684,7 @@
while ((trans = strsep(&val, ","))) {
trans = ast_skip_blanks(trans);
- if (!strncasecmp(trans, "udp", 3))
+ if (!strncasecmp(trans, "udp", 3))
peer->transports |= SIP_TRANSPORT_UDP;
else if (!strncasecmp(trans, "tcp", 3))
peer->transports |= SIP_TRANSPORT_TCP;
@@ -22622,9 +22693,8 @@
else
ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
- if (!peer->socket.type) { /*!< The first transport listed should be used for outgoing */
- peer->socket.type = peer->transports;
- peer->socket.fd = -1;
+ if (!peer->default_outbound_transport) { /*!< The first transport listed should be default outbound */
+ peer->default_outbound_transport = peer->transports;
}
}
} else if (realtime && !strcasecmp(v->name, "regseconds")) {
@@ -22926,12 +22996,21 @@
}
}
- if (!peer->socket.type) {
+ if (!peer->default_outbound_transport) {
/* Set default set of transports */
peer->transports = default_transports;
/* Set default primary transport */
- peer->socket.type = default_primary_transport;
- peer->socket.fd = -1;
+ peer->default_outbound_transport = default_primary_transport;
+ }
+
+ /* The default transport type set during build_peer should only replace the socket.type when...
+ * 1. Registration is not present and the socket.type and default transport types are different.
+ * 2. The socket.type is not an acceptable transport type after rebuilding peer.
+ * 3. The socket.type is not set yet. */
+ if (((peer->socket.type != peer->default_outbound_transport) && (peer->expire == -1)) ||
+ !(peer->socket.type & peer->transports) || !(peer->socket.type)) {
+
+ set_peer_transport(peer, peer->default_outbound_transport);
}
if (fullcontact->used > 0) {
@@ -22956,7 +23035,7 @@
if ((params = strchr(_srvlookup, ';'))) {
*params++ = '\0';
}
-
+
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(peer->socket.type));
if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup ? transport : NULL)) {
Modified: branches/1.6.2/configs/sip.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.2/configs/sip.conf.sample?view=diff&rev=196452&r1=196451&r2=196452
==============================================================================
--- branches/1.6.2/configs/sip.conf.sample (original)
+++ branches/1.6.2/configs/sip.conf.sample Fri May 22 16:59:31 2009
@@ -874,11 +874,14 @@
;remotesecret=guessit ; Our password to their service
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
-;fromdomain=provider.sip.domain
+;fromdomain=provider.sip.domain
;host=box.provider.com
-;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
-; ; accept both tcp and udp. Default is udp. The first transport
-; ; listed will always be used for outgoing connections.
+;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
+; ; accept both tcp and udp. The default transport type is only used for
+; ; outbound messages until a Registration takes place. During the
+; ; peer Registration the transport type may change to another supported
+; ; type if the peer requests so.
+
;usereqphone=yes ; This provider requires ";user=phone" on URI
;callcounter=yes ; Enable call counter
;busylevel=2 ; Signal busy at 2 or more calls
More information about the asterisk-commits
mailing list